1. Technical Field
The present invention relates generally to communication networks, and more particularly, to Session Initiation Protocol networks for carrying calls over a data packet network.
2. Related Art
Traditional telephone networks including the Public Switched Telephone Network (PSTN) and Signaling System Number 7 (SS7) networks have provided closed systems that enabled users to achieve added capabilities beyond merely connecting a call. Initially, message storage capabilities such as those provided by answering machines and voice mail services (in SS7 networks) were popular. Since then, many other services have been developed as SS7 and other intelligent networks (IN and AIN) have gained widespread popularity. With the advent of Internet telephony, a need to provide voice and video mail services as well as traditional services has been realized.
During the past few years, Internet telephony has evolved from being a novelty for the technically oriented seeking party conversation material to a technology that, in the not too distant future, may largely replace the existing telephone networks. Supporting the widespread use of Internet telephony requires a host of standardized protocols to ensure transport audio and video data having a specified quality of service (QoS). These protocols are also needed to provide directory services and to enable signaling. Signaling protocols are of particular interest because they are the basis for advanced services such as mobility, universal numbers, multiparty conferencing, voice mail, and automatic call distribution.
Two signaling protocols that are being used to develop Internet telephony are the ITU-T H.323 suite of protocols and the Session Initiation Protocol (SIP) being developed by the Internet Engineering Task Force (IETF). SIP's strengths include its simplicity, scalability, extensibility, and modularity. As a result, increasing interest in SIP is being realized as the SIP standards and protocol requirements develope into maturity. The SIP Internet Standard proposal is documented in RFC 2543. RFC 2543 defines the setting up, control and tearing down sessions in the Internet. The defined SIP sessions include, but are not limited to, Internet telephone calls and multimedia conferences. They also define protocol requirements for Internet conferencing, telephony, presence, events notification and instant messaging.
Each SIP session may include different types of data including real time data such as audio and video. Currently, however, most of the SIP extensions address audio communication. As a traditional text-based Internet protocol, SIP resembles the hypertext transfer protocol (HTTP) and simple mail transfer protocol (SMTP). SIP uses Session Description Protocol (SDP) for media description. SIP, however, is independent of the packet layer. The protocol is an open standard and is scalable. Among SIP basic features, the protocol also enables personal mobility by providing the capability to reach a called party at a single, location-independent address.
SIP's basic architecture is client/server in nature. The main entities in SIP are the User Agent, the SIP Proxy Server, the SIP Redirect Server and the Registrar. In contrast to H.323, both SIP and H.323 support call routing, call signaling, capabilities exchange, media control, and supplementary services. SIP, however, in the view of many, is more scalable and provides greater flexibility and ease of implementation when building complex systems. H.323, on the other hand, has been widely used because of its manageability, reliability and inter-operability with the PSTN.
The User Agents, or SIP endpoints, function as clients (UACs) when initiating requests and as servers (UASs) when responding to requests. User Agents communicate with other User Agents directly or via an intermediate server. The User Agent also stores and manages call states.
SIP intermediate servers have the capability to behave as proxy or redirect servers. SIP Proxy Servers forward requests from the User Agent to the next SIP server, User Agent within the network and also retain information for billing/accounting purposes. SIP Redirect Servers respond to client requests and inform them of the requested server's address. Numerous hops can take place until reaching the final destination. SIP's tremendous flexibility allows the servers to contact external location servers to determine user or routing policies, and therefore, does not bind the user into only one scheme to locate users. In addition, to maintain scalability, the SIP servers can either maintain state information or forward requests in a stateless fashion.
The SIP registrar is the third entity of a SIP network. The SIP Registrar stores the registration information in a location service via a non-SIP protocol that is received in a registration message sent by the User Agent. Once the information is stored, the Registrar sends the appropriate response back to the user agent.
Generally, SIP is independent of the packet layer and only requires a datagram service. While SIP typically is used over UDP or TCP, it could, without technical changes, be run over IPX, frame relay, ATM AAL5 or X.25. SIP is designed to have a simple architecture and implementation. SIP's simplicity, however, doesn't compromise its power. It may also support encryption and authentication. SIP's client/server orientation offers a level of server-based call management missing in the peer call model that most H.323 endpoints use. In operation, the first thing a SIP client does is to locate a server, typically via DNS. SIP proxies can be easily integrated into firewalls and Network Address Translators. Proposed SIP extensions include specs for user-based call-security management and QoS requirements, as well as the signaling of changes in network conditions.
SIP provides the necessary protocol mechanisms so that end systems and proxy servers can provide call forwarding services, including the equivalent of toll free and 1-900-type feature calls; call-forwarding, no answer; call-forwarding busy; call-forwarding unconditional; called party and calling party number delivery; personal mobility, i.e., the ability to reach a called party under a single, location-independent address even when the user changes terminals. SIP further supports terminal-type negotiation and selection. More specifically, a caller may choose the method for reaching a called party, e.g., via Internet telephony, mobile phone, an answering service, etc.
As may be seen therefore, SIP is a new protocol that provides great promise and expanded capabilities for future generations of telephone systems. Because the SIP protocol requirements are in development, however, certain capabilities and features have not been provided for or defined.
One particular capability that is not being provided but that the present inventors believe to be necessary is that of call monitoring by the various governmental agencies under specified and court approved circumstances in order to combat criminals, terrorists and rascals of all types. More particularly, traditional wiretapping has been effective only when utilized secretly. The traditional PSTN and SS7 networks made anonymous wiretapping an easy feat to accomplish. With data packet networks, such as SIP, anonymous “wiretapping” or call monitoring is not so easy. One reason is that data packets carrying the communication to be monitored are likely to be transmitted through a plurality of nodes by any one of a large plurality of paths. Thus, the potential of data packets forming a communication being transmitted by way of different routes makes it difficult to monitor the communication. If all the packets that form a communication that is to be monitored are routed through a single node that contains monitoring equipment, however, call monitoring becomes detectable because the path defined by the headers appended to the packet would reveal the single node with the monitoring equipment.
An alternative design, of course, is to place call monitoring equipment at every node or at least at every gateway into the data packet network. This solution however, is problematic because of cost and because of the lack of control that a governmental agency might encounter with such widespread distribution of the call monitoring equipment. What is needed, therefore, is a system and process for monitoring select calls in a data packet network that avoids detection, provides adequate control, and is not too expensive.